linux/sound/soc/codecs/ssm2602.c

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// SPDX-License-Identifier: GPL-2.0-or-later
//
// File: sound/soc/codecs/ssm2602.c
// Author: Cliff Cai <Cliff.Cai@analog.com>
//
// Created: Tue June 06 2008
// Description: Driver for ssm2602 sound chip
//
// Modified:
// Copyright 2008 Analog Devices Inc.
//
// Bugs: Enter bugs at http://blackfin.uclinux.org/
#include <linux/delay.h>
#include <linux/module.h>
#include <linux/regmap.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 16:04:11 +08:00
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include "ssm2602.h"
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
const struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct regmap *regmap;
enum ssm2602_type type;
unsigned int clk_out_pwr;
};
/*
* ssm2602 register cache
* We can't read the ssm2602 register space when we are
* using 2 wire for device control, so we cache them instead.
* There is no point in caching the reset register
*/
static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = {
{ .reg = 0x00, .def = 0x0097 },
{ .reg = 0x01, .def = 0x0097 },
{ .reg = 0x02, .def = 0x0079 },
{ .reg = 0x03, .def = 0x0079 },
{ .reg = 0x04, .def = 0x000a },
{ .reg = 0x05, .def = 0x0008 },
{ .reg = 0x06, .def = 0x009f },
{ .reg = 0x07, .def = 0x000a },
{ .reg = 0x08, .def = 0x0000 },
{ .reg = 0x09, .def = 0x0000 }
};
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
"Line", "Mic",
};
static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum ssm2602_enum[] = {
SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
ssm2602_input_select),
SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
ssm2602_deemph),
};
static const DECLARE_TLV_DB_RANGE(ssm260x_outmix_tlv,
0, 47, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
48, 127, TLV_DB_SCALE_ITEM(-7400, 100, 0)
);
static const DECLARE_TLV_DB_SCALE(ssm260x_inpga_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(ssm260x_sidetone_tlv, -1500, 300, 0);
static const struct snd_kcontrol_new ssm260x_snd_controls[] = {
SOC_DOUBLE_R_TLV("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 45, 0,
ssm260x_inpga_tlv),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
};
static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
0, 127, 0, ssm260x_outmix_tlv),
SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
7, 1, 0),
SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1,
ssm260x_sidetone_tlv),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
};
/* Output Mixer */
static const struct snd_kcontrol_new ssm260x_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
};
static const struct snd_kcontrol_new mic_ctl =
SOC_DAPM_SINGLE("Switch", SSM2602_APANA, 1, 1, 1);
/* Input mux */
static const struct snd_kcontrol_new ssm2602_input_mux_controls =
SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
static int ssm2602_mic_switch_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
/*
* According to the ssm2603 data sheet (control register sequencing),
* the digital core should be activated only after all necessary bits
* in the power register are enabled, and a delay determined by the
* decoupling capacitor on the VMID pin has passed. If the digital core
* is activated too early, or even before the ADC is powered up, audible
* artifacts appear at the beginning and end of the recorded signal.
*
* In practice, audible artifacts disappear well over 500 ms.
*/
msleep(500);
return 0;
}
static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("LLINEIN"),
};
static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
ssm260x_output_mixer_controls,
ARRAY_SIZE(ssm260x_output_mixer_controls)),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
SND_SOC_DAPM_SWITCH_E("Mic Switch", SSM2602_APANA, 1, 1, &mic_ctl,
ssm2602_mic_switch_event, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_INPUT("MICIN"),
};
static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
ssm260x_output_mixer_controls,
ARRAY_SIZE(ssm260x_output_mixer_controls) - 1), /* Last element is the mic */
};
static const struct snd_soc_dapm_route ssm260x_routes[] = {
{"DAC", NULL, "Digital Core Power"},
{"ADC", NULL, "Digital Core Power"},
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"ROUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
{"Line Input", NULL, "LLINEIN"},
{"Line Input", NULL, "RLINEIN"},
};
static const struct snd_soc_dapm_route ssm2602_routes[] = {
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
{"RHPOUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"Input Mux", "Line", "Line Input"},
{"Input Mux", "Mic", "Mic Switch"},
{"ADC", NULL, "Input Mux"},
{"Mic Switch", NULL, "Mic Bias"},
{"Mic Bias", NULL, "MICIN"},
};
static const struct snd_soc_dapm_route ssm2604_routes[] = {
{"ADC", NULL, "Line Input"},
};
static const unsigned int ssm2602_rates_12288000[] = {
8000, 16000, 32000, 48000, 96000,
};
static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
.list = ssm2602_rates_12288000,
.count = ARRAY_SIZE(ssm2602_rates_12288000),
};
static const unsigned int ssm2602_rates_11289600[] = {
8000, 11025, 22050, 44100, 88200,
};
static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
.list = ssm2602_rates_11289600,
.count = ARRAY_SIZE(ssm2602_rates_11289600),
};
struct ssm2602_coeff {
u32 mclk;
u32 rate;
u8 srate;
};
#define SSM2602_COEFF_SRATE(sr, bosr, usb) (((sr) << 2) | ((bosr) << 1) | (usb))
/* codec mclk clock coefficients */
static const struct ssm2602_coeff ssm2602_coeff_table[] = {
/* 48k */
{12288000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x0)},
{18432000, 48000, SSM2602_COEFF_SRATE(0x0, 0x1, 0x0)},
{12000000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x1)},
/* 32k */
{12288000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x0)},
{18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)},
{12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)},
/* 16k */
{12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)},
{18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)},
{12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)},
/* 8k */
{12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)},
{18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)},
{11289600, 8000, SSM2602_COEFF_SRATE(0xb, 0x0, 0x0)},
{16934400, 8000, SSM2602_COEFF_SRATE(0xb, 0x1, 0x0)},
{12000000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x1)},
/* 96k */
{12288000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x0)},
{18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)},
{12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)},
/* 11.025k */
{11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)},
{16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)},
{12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)},
/* 22.05k */
{11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)},
{16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)},
{12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)},
/* 44.1k */
{11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)},
{16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)},
{12000000, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x1)},
/* 88.2k */
{11289600, 88200, SSM2602_COEFF_SRATE(0xf, 0x0, 0x0)},
{16934400, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x0)},
{12000000, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x1)},
};
static inline int ssm2602_get_coeff(int mclk, int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(ssm2602_coeff_table); i++) {
if (ssm2602_coeff_table[i].rate == rate &&
ssm2602_coeff_table[i].mclk == mclk)
return ssm2602_coeff_table[i].srate;
}
return -EINVAL;
}
static int ssm2602_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
unsigned int iface;
if (srate < 0)
return srate;
regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
switch (params_width(params)) {
case 16:
iface = 0x0;
break;
case 20:
iface = 0x4;
break;
case 24:
iface = 0x8;
break;
case 32:
iface = 0xc;
break;
default:
return -EINVAL;
}
regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
IFACE_AUDIO_DATA_LEN, iface);
return 0;
}
static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
if (ssm2602->sysclk_constraints) {
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
ssm2602->sysclk_constraints);
}
return 0;
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(dai->component);
if (mute)
regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE,
APDIGI_ENABLE_DAC_MUTE);
else
regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE, 0);
return 0;
}
static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_component *component = codec_dai->component;
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
if (dir == SND_SOC_CLOCK_IN) {
if (clk_id != SSM2602_SYSCLK)
return -EINVAL;
switch (freq) {
case 12288000:
case 18432000:
ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
break;
case 11289600:
case 16934400:
ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
break;
case 12000000:
ssm2602->sysclk_constraints = NULL;
break;
default:
return -EINVAL;
}
ssm2602->sysclk = freq;
} else {
unsigned int mask;
switch (clk_id) {
case SSM2602_CLK_CLKOUT:
mask = PWR_CLK_OUT_PDN;
break;
case SSM2602_CLK_XTO:
mask = PWR_OSC_PDN;
break;
default:
return -EINVAL;
}
if (freq == 0)
ssm2602->clk_out_pwr |= mask;
else
ssm2602->clk_out_pwr &= ~mask;
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
}
return 0;
}
static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(codec_dai->component);
unsigned int iface = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface |= 0x0040;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0003;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
iface |= 0x0090;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x0010;
break;
default:
return -EINVAL;
}
/* set iface */
regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
return 0;
}
static int ssm2602_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid on, osc and clkout on if enabled */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
ssm2602->clk_out_pwr);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
PWR_CLK_OUT_PDN | PWR_OSC_PDN);
break;
case SND_SOC_BIAS_OFF:
/* everything off */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF, PWR_POWER_OFF);
break;
}
return 0;
}
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000)
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops ssm2602_dai_ops = {
.startup = ssm2602_startup,
.hw_params = ssm2602_hw_params,
.digital_mute = ssm2602_mute,
.set_sysclk = ssm2602_set_dai_sysclk,
.set_fmt = ssm2602_set_dai_fmt,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver ssm2602_dai = {
.name = "ssm2602-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SSM2602_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SSM2602_FORMATS,},
.ops = &ssm2602_dai_ops,
.symmetric_rates = 1,
.symmetric_samplebits = 1,
};
static int ssm2602_resume(struct snd_soc_component *component)
{
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
regcache_sync(ssm2602->regmap);
return 0;
}
static int ssm2602_component_probe(struct snd_soc_component *component)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
int ret;
regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
ret = snd_soc_add_component_controls(component, ssm2602_snd_controls,
ARRAY_SIZE(ssm2602_snd_controls));
if (ret)
return ret;
ret = snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
ARRAY_SIZE(ssm2602_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, ssm2602_routes,
ARRAY_SIZE(ssm2602_routes));
}
static int ssm2604_component_probe(struct snd_soc_component *component)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
int ret;
ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
ARRAY_SIZE(ssm2604_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, ssm2604_routes,
ARRAY_SIZE(ssm2604_routes));
}
static int ssm260x_component_probe(struct snd_soc_component *component)
{
struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
int ret;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
if (ret < 0) {
dev_err(component->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
/* set the update bits */
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
/*select Line in as default input*/
regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
switch (ssm2602->type) {
case SSM2602:
ret = ssm2602_component_probe(component);
break;
case SSM2604:
ret = ssm2604_component_probe(component);
break;
}
return ret;
}
static const struct snd_soc_component_driver soc_component_dev_ssm2602 = {
.probe = ssm260x_component_probe,
.resume = ssm2602_resume,
.set_bias_level = ssm2602_set_bias_level,
.controls = ssm260x_snd_controls,
.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
.dapm_widgets = ssm260x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ssm260x_dapm_widgets),
.dapm_routes = ssm260x_routes,
.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
};
static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
{
return reg == SSM2602_RESET;
}
const struct regmap_config ssm2602_regmap_config = {
.val_bits = 9,
.reg_bits = 7,
.max_register = SSM2602_RESET,
.volatile_reg = ssm2602_register_volatile,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = ssm2602_reg,
.num_reg_defaults = ARRAY_SIZE(ssm2602_reg),
};
EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
int ssm2602_probe(struct device *dev, enum ssm2602_type type,
struct regmap *regmap)
{
struct ssm2602_priv *ssm2602;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
if (ssm2602 == NULL)
return -ENOMEM;
dev_set_drvdata(dev, ssm2602);
ssm2602->type = type;
ssm2602->regmap = regmap;
return devm_snd_soc_register_component(dev, &soc_component_dev_ssm2602,
&ssm2602_dai, 1);
}
EXPORT_SYMBOL_GPL(ssm2602_probe);
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");